Re: [asterisk-users] Need ISDN call generator
Keep an eye out for older model INET Spectra call generators, with ISDN / SS7 stacks. These days the old boxes are being sold off very cheaply on popular auction sites. Hammer was the other popular call generator hardware that you might find being sold at a fraction of the original cost. HTH Darren On 28 August 2016 at 10:20, Hooman Fazaeli <hoomanfaza...@gmail.com> wrote: > > Hi > > To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk > system, > we are looking to buy an ISDN call generator/simulator device. > > The minimum requirements include: > > - Not too expensive > - PRI support (BRI support is a plus) > - CCS+CRC4 farming + HDB3 coding > - EuroISDN (DSS1) support. > - A minimum of 4 ports (120 channels/concurrent calls) > - Compatibility with Digium cards. > - DUT in TE mode. > - Reliable & stable operation. > > I would like to hear your recommendations for and experiences about > such a device. Recommendations on hand crafted systems using > Asterisk, DAHDI and PRI cards on any OS which has worked stably for > someone are also welcome > > Thanks. > > > -- > Best regards > Hooman Fazaeli > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
On Mar 22, 2013, at 5:22 AM, Florian Wolters flor...@florian-wolters.de wrote: So I did setup another Extension leading me to a MeetMe conference to at least listen to some MoH while waiting for the 15 Minutes to exceed. This showed the same behaviour. After exactly 15 Minutes, the call is terminated - namely by the provider. The analysis of the dump in Wireshark shows the last 6 SIP packets: 2013-03-21 15:56:50.648141217.0.17.170 = 172.16.0.2Request: INVITE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.648325172.16.0.2 = 217.0.17.170 Status: 100 Trying 2013-03-21 15:56:50.648427172.16.0.2 = 217.0.17.170 Status: 200 OK, with session description 2013-03-21 15:56:50.731436217.0.17.170 = 172.16.0.2Request: ACK sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735426217.0.17.170 = 172.16.0.2Request: BYE sip:02341234567890@79.253.136.186:5060 2013-03-21 15:56:50.735590172.16.0.2 = 217.0.17.170 Status: 200 OK (manually copied that from the Wireshark window). This looks to me as if the provider for some reason does an INVITE after 15 Minutes, that is not correctly handled by my Asterisk. Is there any timer inside the SIP protocol, that may be aged by 15 Minutes? Or should I have a deeper look at the SIP packets? Sip session timers? http://doxygen.asterisk.org/trunk/sip_session_timers.html -d -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
both would be appreciated. if you can send me a backtrace, that'd be great On Jun 22, 2012, at 8:06 PM, Jeremy Kister wrote: On 6/20/2012 8:24 AM, Darren Sessions wrote: I just finished replying to your direct email (which you can disregard now as this seems to be a different problem). I'm pretty sure I know what the issue is, but I'll have to get back to you later this evening (my time). I have a different problem- i just compiled app_swift 3 from the new git repo for asterisk 1.8.13.0 asterisk loads the module fine, but as soon as i try to swift anything, asterisk core dumps. i'll be glad to post the corefile or sample extensions.conf if desired. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
Hi Jakob, I just finished replying to your direct email (which you can disregard now as this seems to be a different problem). I'm pretty sure I know what the issue is, but I'll have to get back to you later this evening (my time). - D On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger wrote: Hi, i am trying to install the just from git cloned app_swift version. Compiling works fine. Install as well. But if i try to load the module at Asterisk it fails with. Command 'module load app_swift.so ' failed. [Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module: Error loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close [Jun 20 11:29:51] WARNING[24217]: loader.c:850 load_resource: Module 'app_swift.so' could not be loaded. My System Informations: server*CLI core show version Asterisk 1.8.13.0 built by root @ server on a x86_64 running Linux on 2012-06-20 08:55:14 UTC root@server:~# uname -r 3.2.0-25-generic root@server:~# ldd /usr/lib/asterisk/modules/app_swift.so linux-vdso.so.1 = (0x7fff6d3ff000) libc.so.6 = /lib/x86_64-linux-gnu/libc.so.6 (0x7f2010a65000) /lib64/ld-linux-x86-64.so.2 (0x7f2011041000) root@server:~# cat /etc/ld.so.conf.d/swift.conf /opt/swift/lib root@server:~#ldconfig -v | grep swift /opt/swift/lib: libswift.so.6 - libswift.so.6.0 libceplex_de.so.6 - libceplex_de.so.6.0 libceplang_de.so.6 - libceplang_de.so.6.0 root@server:~# swift -V Cepstral Swift v6.0.1, March 2012 Default Voice: Matthias-8kHzv6.0.0 Language: German v5.1.0 Lexicon:unknown v0.0.0 Concurrency:1 Port(s) Registered 0 Port(s) In Use Distribution: No audio distribution license was found. Saving audio to a file is disabled. Copyright (C) 2000-20012, Cepstral LLC. Do You have any Ideas why that won't work? Best Regards Jakob Böttger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift beta release
Hi folks, Just a note to let everyone know I've finally finished up the new BETA release of app_swift (now v3.0.1 b1). This release introduces some pretty major changes to app_swift such as: - The entire code-base has now been unified and the build system auto detects which Asterisk version you're using (yay! one branch!) - Auto-detection and support for both the Cepstral 5.0 and 6.0 engines - Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10 - Asterisk 1.2 support has been dropped. I have only been able to do some basic testing with all these permutations of Asterisk and the Cepstral engines on a few of my machines here at the house and need some volunteers to help out and be guinea-pigs. Please email me directly with any feedback you might have. I've updated my github repo with the new app_swift code which can be downloaded using git. git clone git://github.com/dmsessions/app_swift.git Thanks, - D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - T.38 unreliable on a LAN : truth or obscurantism ?
T.38 is tolerant of most network conditions, ... the challenges in getting reliable performance are usually limited to getting the interop right once, but the absolute success rate will depend on the quality of your T.38/PSTN gateway's fax implementation. In general terms, T.38 is actually the right way to cope with lossy or high jitter network conditions, and so it's reliable over most networks. The question people usually ask is whether fax over G.711 is unreliable on a LAN. To which the answer would be a definite 'it depends' ;-) -d On Feb 15, 2012, at 3:03 PM, Olivier wrote: Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Comments ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - T.38 unreliable on a LAN : truth or obscurantism ?
On Feb 15, 2012, at 4:03 PM, Olivier wrote: 2012/2/15, Darren Nickerson darren.nicker...@ifax.com: T.38 is tolerant of most network conditions, ... the challenges in getting reliable performance are usually limited to getting the interop right once, but the absolute success rate will depend on the quality of your T.38/PSTN gateway's fax implementation. In general terms, T.38 is actually the right way to cope with lossy or high jitter network conditions, and so it's reliable over most networks. Yes. An other thing to factor in, is how Asterisk's load could influence its capability to let faxes passing through. To me, if Asterisk is installed on a modern CPU (dual core and more) and is configured in such a way that no transcoding happen, then passing faxes through is easy and works reliably. Opinions ? The devil is in the details, but in general it's nowhere near that simple. You don't clarify what pass-through role Asterisk is playing here. G.711? T.38? What are you passing through TO? A TDM card connected to the PSTN? Or some SIP trunking provider, who themselves may be using G.711 or T.38 ... Assuming you mean the specific case of one local LAN hop over SIP, connecting directly to a well-configured PSTN card on the same Asterisk server, it's possible to get reliable faxing over G.711 with careful network configuration, good and well configured ethernet interfaces, correct jitter buffer, gain and echo cancelation settings, etc etc. -d -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - T.38 unreliable on a LAN : truth or obscurantism ?
On Feb 15, 2012, at 3:49 PM, Olivier wrote: 2012/2/15, Tim Nelson tnel...@rockbochs.com: - Original Message - Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Is T.38 actually in use in this scenario? Or are you simply passing the fax call through Asterisk as 'normal' audio (G.711u/a, etc)? Yes, T.38 is in use between each gateway and Asterisk (I should have specified this more clearly) : Fax Machine -- Analog Gw --T.38-- Asterisk --T.38 -- Digital Gw --ISDN-- PSTN Assuming you have Asterisk doing T.38 pass-through here, reinviting the T.38 payload to go directly between the analog GW and the Digital GW, and assuming that 'Digital Gw' has a good T.30 fax engine inside of it (because after all, the gateway is what's speaking convention audio-based fax to the remote sender/receiver, the above setup should work well independent of network conditions. T.38 has ways of coping with extremely bad connections (via packet redundancy or FEC error correction) that you probably would not need on a LAN. Note, however, the use of T.38 versus G.711 may limit the speed of your faxing to 14,400 and prevent the fax protocol from using its own error correction (many T.38 gateway implementations wrong-headedly disable ECM error correction). When it comes to faxing over a LAN, the choice of T.38 versus G.711 uLaw/aLaw is less than obvious. In your case, it will be highly dependent upon each piece of your call flow. The fax machine, the analog gateway, how Asterisk is setup, the digital gateway and the quality of the PSTN line. These days you cannot trust that your PSTN carrier is using TDM routes, sometimes they slip a little T.38 in the middle on you, and all bets are off. No matter what scenario you go with though, you probably want to get Asterisk out of the media path and get a gateway-to-gateway conversation going eventually. -Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift tts module - new home.
Hi Folks, After receiving a surprising amount of emails from Asterisk community members, I thought I'd fire something off to the users list to clear any confusion regarding the Asterisk Forge (forge.asterisk.org) website and the future of the app_swift text-to-speech module. With regards to the Asterisk Forge website redirecting to GitHub, this has been a long time coming. Emails were sent out to the various lists warning folks that the hosted GForge site was going away - so no one should be too surprised - 'nuf said there. As far as the app_swift project is concerned, with the exception of moving things around as far as location, it is business as usual. The app_swift code for *all* the different versions of Asterisk is now being hosted on GitHub at https://github.com/dmsessions/app_swift . This is a good thing and will make life easier. btw, I love git. If you don't yet, you will . . someday soon . . Individual tar files for each of the different versions of app_swift, which is what 99% of people are going to want, are all available for download on my website at http://www.darrensessions.com by clicking the 'Downloads' button at the very top of the page. That is all my friends. Seasons Greetings! - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best VoIP conferencing phone ?
We've been happy with the polycom IP 7000. Darren Wiebe On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote: Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue *1)Polycom SoundStation IP 7000 * *Why it's best: *The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20’ 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. * * *2) Polycom Voicestation 500* * * *Why it's a best pick: *The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7’ 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. * * *3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S* * * *Why it's a best pick: *With a 360 10’ radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. * * *4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone* * * *Why it's a best pick: *The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. * * *5)Polycom SoundStation VTX 1000* * * *Why it's a best pick: *The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20’ 360 radius. 6)Polycom® SoundStation® IP 5000* 7) GXP2120 6-line Executive HD IP Phone* On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. ** ** Regards, ** ** Faisal Hanif ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, November 30, 2011 11:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind *Subject:* [asterisk-users] Best VoIP conferencing phone ? ** ** Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift for Asterisk 10
Hey there folks, I'd sent this to the list last night and got reject email this morning. Apparently it is always a good idea to have an active subscription to the list you are trying to post to - just one of those things. :) In any case, a new beta version of app_swift is available for Asterisk 10. I put it up in the Asterisk Forge on the 25th of last month, but wanted to wait to post something on the users list until I'd had a chance to really test it a bit (so far so good). http://forge.asterisk.org/gf/project/app_swift/frs/ I have to say, the combination of Asterisk 10 and this latest version of app_swift is absolutely the best sounding of any release to-date! I've been *very* impressed so far. Also, just fyi . . there are some extremely minor tweaks I'll be back-porting to the other app_swift versions shortly. I hope to get that done this weekend or next depending on my free time. Enjoy, - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk
Why not firewall hack attempts after 3 tries? When we started doing that the quantity of hacking attempts dropped right off. We also setup our own fail2ban sharing server so that we could share the bans across multiple servers. Have a look at http://www.f2bshare.org/index.php?title=Main_Page if you want to do something similar. Why try to make Asterisk into something it's not intended to be? Just use your firewall for what it's good at. -- Darren Wiebe On 7/23/11 11:38 AM, CDR wrote: I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user information, and also, b) disable any response to any REGISTER packet altogether. Can somebody please write patch? Or should we go broke trying to stop the flood of criminals coming from abroad? Federico On Sat, Jul 23, 2011 at 1:00 PM, asterisk-users-requ...@lists.digium.com wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: use dahdi for local terminal modem access? (Lyle Giese) 2. dialplan pattern help (Armand Fumal) 3. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined (Patrick Lists) 4. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined (Paul Belanger) -- Message: 1 Date: Sat, 23 Jul 2011 09:29:26 -0500 From: Lyle Giesel...@lcrcomputer.net Subject: Re: [asterisk-users] use dahdi for local terminal modem access? To: asterisk-users@lists.digium.com Message-ID:4e2adac6.4010...@lcrcomputer.net Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 07/22/11 22:47, William Stillwell wrote: Um, no VOIP involved here. Wrong. What do you think Asterisk is? Chopped meat? It's a VoIP switch. All traffic inside Asterisk is VoIP. I have an asterisk server with 2 23B+D PRI's I want to telnet/ssh into the asterisk server, and make an outbound call serial based modem/terminal connection (Like the 80/90's BBS Days). No TCP/IP or PPP or crazyness (ie, dialing into a Modem set to AA hooked to a Cisco Console Port) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Friday, July 22, 2011 8:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use dahdi for local terminal modem access? On 07/22/11 18:13, William Stillwell wrote: I have some terminals that have phone lines. One of my tech had an idea of using IAXmodem or something similar to use existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console. Anybody ever heard of doing this? I would think maybe would use iaxmodem maybe and a shell terminal app? (basically I'm dialing into a remote access device that uses a pots like for remote administration, and don't want to string a channel bank off my asterisk box, and a hook to a modem) -- Depends on your expectation. Because of compression in the codecs, it will be hard to get fast dialup. If you mean ssh or telnet, it might work. If you mean vnc or RDP over this, you may not get enough usable bandwidth to do that. Given this, I have in an emergency dialed into a RAS server via a VoIP line. My laptop connected at 14,400bps. All I needed to do was telnet into an APC masterswitch to toggle power on one outlet. It worked. I was surprised at getting a 14,400bps connect. I was not expecting that high and really did not need that high. 300 baud probably would have been fast enough to telnet into an APC masterswitch. Lyle Giese LCR Computer Services, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
[asterisk-users] Sharing Fail2ban data
Good Day, I've been doing a little work that I wanted to share. We've had a number of Asterisk systems that have been under heavier than normal attack. We use fail2ban but we either have to let each system be exposed or keep all the data synchronized which is a bit of a chore. I wrote a little server that assists in keeping data synchronized across sites. If you're interested in using it to assist in managing your own fail2ban sharing list I'll gladly share it. I also am offering it as a free service for those who are interested in contributing to a blacklist. If you're interested the information is available here: http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in the server code just drop me an email. Darren Wiebe dar...@aleph-com.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balance and Failover
You could use a sip proxy front end like Kamailio. Sent from my iPhone On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote: Hi All Does anyone know about any tool that does to Asterisk what mod_jk does for JBoss/Tomcat: a load-balance/failover server that is constantly connected to Asterisk backend servers and is capable of identify loaded or down servers? Regards Antônio Theóphilo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality
Well, the downside to wav files is the disk i/o. Asterisk will and does translate the audio frames from ulaw to whatever other codec. Sent from my iPhone On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
We recently completed a project using products from here: http://www.controlbyweb.com/webrelay/ They were easy to setup and can be controlled in a variety of fashions included http queries. Darren Wiebe On 18/10/2010 8:34 AM, Marco Signorini wrote: Hi Did you looked at Arduino + Ethernet Shield? Is something you can program in C or C++ to receive a simple TCP and/or HTTP packet and turn on an external relay. From the dialplan you can run an http query through curl and/or an external AGI command. Best regards, Marco Signorini. -- Marco Signorini http://www.ethermania.com http://www.ingegnitech.com Roberto Piola wrote: we're using a Damocles Mini (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of course, the damocles will have to drive a high-power relay. the damocles can be driven via snmp, so you'll have to simply call the snmpset unix standard utility On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades list-aster...@skycomuk.com wrote: Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only control 750W so you will probably need to get it to control a more powerfull relay as a heater is going to take a lot of current. It can be controlled by a virtual serial port so you just program the extension to make a system() call to a simple script which sends a string of characters to the serial port. That device is quite expensive. You could probably find something much cheaper on ebay. Gilles wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift for Asterisk 1.8
Just thought I'd let everyone know I've got a new beta version of app_swift up for Asterisk 1.8 on http://forge.asterisk.org. - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote: On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins d...@cognation.net wrote: Any thoughts on why the lack of traffic? Cheers, Dean Not enough applications to play immature bathroom sounds. You could well be right, but consider for a moment a few alternatives. Perhaps it's the $5000 up front just to be listed? I see the fee's reduced to $2500 now as a promo, but still that's a huge barrier for most. Or perhaps its the fact that the nature of the apps that get listed means they aren't usually 'purchase-able' with a simple 'click to buy' (how do you sell SIP trunking with a click-to-buy???) - and as a consequence there's no purchase capability built into the asteriskexchange site, just link outs to different purchase-ish URLs for the various products. Anyone looking to sell their app would need to develop their own point-of-sale/payment processing systems so it's really not an 'app store' at all in the traditional sense. Kudos to digium for realizing this goal, but I think the $5000 get-in cost has resulted in the lack of interest/popularity, and limited the listings to only the largest, most profitable asterisk/digium partners. -d -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift v2.0 released
Hi all, Thought I'd mention that the new version of the app_swift text-to-speech module for Asterisk 1.2, 1.4, and 1.6 has been released at it's new home on the Asterisk Forge. http://forge.asterisk.org/gf/project/app_swift/ For those that are unaware, app_swift provides a direct interface with the Cepstral text-to-speech engine so instead of having to call the Cepstral engine and write then read an audio file (i.e. disk I/O), you can call the library directly and stream the audio straight to the Asterisk channel. Additionally, the app_swift module supports DTMF detection with a max digits and timeout value as well (similar to the AGI get data functionality). The new version of app_swift has been built and tested on the latest releases of Asterisk for each of their respective code-bases (1.2.40, 1.4.32, and 1.6.2.8) using the Cepstral 5.x libraries. Any questions or feedback, please let me know. Thanks, - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over running my did's
On 10/04/2010 9:24 PM, Timothy C Litwiller wrote: I have a did with 20 channels from didforsale. that we use to let local members call to listen to a conference several times a week without long distance charges. The upcoming call is getting more interest than usual and from places that are not local so we want to use a free conference service in addition to the local conference. How can I setup a conference on my asterisk box for the people that normally call in there and also call an outside number for those that are above and beyond the 20 lines channels I can provide and the are long distance anyways so a number here or a number in iowa doesn't make them any difference. is there a way that I could call the outside conference # and then transfer it to a local asterisk conference and then hang up can call the local asterisk conference back - and if I do that how do I hang up the long distance conference when it is done? I seem to be missing some basic understanding here. I would call into the free conference service and then transfer that call into my meetme conference. If you're using Trixbox you can use the MeetMe web control to disconnect the call when you're done. You can also disconnect calls from the asterisk cli using the soft hangup command. Darren Wiebe dar...@aleph-com.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt posing as digium
On Mar 10, 2010, at 5:35 PM, Thomas Kenyon wrote: Did anyone else just get what looks like a phising attempt pretending to be from digium? It appears to be full of links to http://app.en25.com/e/er.aspx I must admit, it looks genuine. I suspect you'll find it _IS_ genuine. The en25.com server is an Eloqua box, ... that's the CRM technology Digium uses to track their campaigns. -Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
Well, never mind on this (didn't get any responses anyways). I basically
removed the meetme announcement options and wrote the functionality from
scratch into my AGI framework along with an announcement queuing daemon that
runs continuously every second in the background that generates a call file and
plays back the user name recording. Hasn't added any CPU overhead with the call
processing and along with working as intended I think there maybe some other
unique capabilities for it down the road.
In any case, thought I'd update the thread.
Cheers,
- Darren
On Jan 11, 2010, at 10:05 AM, Darren Sessions wrote:
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference = 1.
I can see where the meetme.c app actually processes it using the
ast_pthread_create_background(conf-announcethread, NULL, announce_thread,
conf); function. The problem is that it's passing the conf data and not the
chan data so it filters out the new caller to the conference and announces
the caller's name to the rest of the conference with the announce_thread
function. Without the chan data available, it makes quick and dirty hacks
even impossible without more insight into the structure of the app ( i was
thinking of just adding a seperate ast_streamfile / ast_waitstream with the
chan variable using an if current-announcetype == CONF_HASJOIN or something
like that).
Unless I'm missing a way to pass the Asterisk API function
ast_pthread_create_background more than one argument and then modify the
announce_thread to accommodate it, I'm at a bit of a loss on a good way of
doing this without making Asterisk seg fault.
The second idea I had was to use a simple conf-background.agi (below) and do
it that way while altering how meetme is called from the actual separate
conferencing agi app. This method does work for announcing the user but the
separate channels refuse to mix audio afterwards (and I have tried every
trick in the book I can think of with this one down to EAGI stuff). If I take
the 'b' option off of the MeetMe call in the AGI script, the audio passes
perfectly. Additionally, attempts at using the manager interface to unlock,
unmute, etc. the conference have no effect. Aside from the audio (obviously a
big deal), the script runs as designed (DTMF detection, etc.).
Any ideas or help would be appreciated.
Many thanks,
- Darren
POI:
Asterisk 1.6.1.6
app_meetme.c - line 1601 (the announce_thread function)
app_meetme.c - line 1817 (the conf_run function)
-- snip --
#!/usr/bin/perl -w
use strict;
use warnings;
use lib '/var/lib/asterisk/agi-bin';
use DBI;
use Asterisk::AGI;
our ($AGI,%v,%ast);
$AGI = new Asterisk::AGI;
%ast = $AGI-ReadParse();
$v{chan} = $ast{channel};
$v{lang} = $AGI-get_variable('CHANNEL(language)');
$v{conf} = $AGI-get_variable('conference_call');
$v{dbh} = sanitized
($v{q},$v{r}) = undef;
$v{q} = SELECT members FROM sanitized WHERE confno = '.$v{conf}.';
$AGI-verbose($v{q});
$v{q} = $v{dbh}-prepare($v{q});
if (!$v{q}-execute) {
exit;
}
$v{r} = $v{q}-fetchrow_hashref();
$v{q}-finish();
$v{dbh}-disconnect;
if ($v{r}{members} 1) {
$AGI-stream_file(/var/spool/asterisk/meetme/meetme-username-.$v{conf}.-.$v{r}{members});
}
while (!$v{loop}) {
exit if (!$AGI-channel_status($v{chan}));
$v{rc} = $AGI-wait_for_digit('6');
}
exit;
--
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[asterisk-users] MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
Hi all,
I'm trying to get the MeetMe system to take a caller and announce to them
they've joined the conference in addition to the other members of the
conference assuming previous members of the conference = 1.
I can see where the meetme.c app actually processes it using the
ast_pthread_create_background(conf-announcethread, NULL, announce_thread,
conf); function. The problem is that it's passing the conf data and not the
chan data so it filters out the new caller to the conference and announces the
caller's name to the rest of the conference with the announce_thread function.
Without the chan data available, it makes quick and dirty hacks even impossible
without more insight into the structure of the app ( i was thinking of just
adding a seperate ast_streamfile / ast_waitstream with the chan variable using
an if current-announcetype == CONF_HASJOIN or something like that).
Unless I'm missing a way to pass the Asterisk API function
ast_pthread_create_background more than one argument and then modify the
announce_thread to accommodate it, I'm at a bit of a loss on a good way of
doing this without making Asterisk seg fault.
The second idea I had was to use a simple conf-background.agi (below) and do it
that way while altering how meetme is called from the actual separate
conferencing agi app. This method does work for announcing the user but the
separate channels refuse to mix audio afterwards (and I have tried every trick
in the book I can think of with this one down to EAGI stuff). If I take the 'b'
option off of the MeetMe call in the AGI script, the audio passes perfectly.
Additionally, attempts at using the manager interface to unlock, unmute, etc.
the conference have no effect. Aside from the audio (obviously a big deal), the
script runs as designed (DTMF detection, etc.).
Any ideas or help would be appreciated.
Many thanks,
- Darren
POI:
Asterisk 1.6.1.6
app_meetme.c - line 1601 (the announce_thread function)
app_meetme.c - line 1817 (the conf_run function)
-- snip --
#!/usr/bin/perl -w
use strict;
use warnings;
use lib '/var/lib/asterisk/agi-bin';
use DBI;
use Asterisk::AGI;
our ($AGI,%v,%ast);
$AGI = new Asterisk::AGI;
%ast = $AGI-ReadParse();
$v{chan} = $ast{channel};
$v{lang} = $AGI-get_variable('CHANNEL(language)');
$v{conf} = $AGI-get_variable('conference_call');
$v{dbh} = sanitized
($v{q},$v{r}) = undef;
$v{q} = SELECT members FROM sanitized WHERE confno = '.$v{conf}.';
$AGI-verbose($v{q});
$v{q} = $v{dbh}-prepare($v{q});
if (!$v{q}-execute) {
exit;
}
$v{r} = $v{q}-fetchrow_hashref();
$v{q}-finish();
$v{dbh}-disconnect;
if ($v{r}{members} 1) {
$AGI-stream_file(/var/spool/asterisk/meetme/meetme-username-.$v{conf}.-.$v{r}{members});
}
while (!$v{loop}) {
exit if (!$AGI-channel_status($v{chan}));
$v{rc} = $AGI-wait_for_digit('6');
}
exit;
--
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Re: [asterisk-users] Best Firewall Suggestions?
Steve Totaro wrote: On Tue, Oct 13, 2009 at 2:41 PM, SIP s...@arcdiv.com mailto:s...@arcdiv.com wrote: David Wathen wrote: Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen Depends on what level of firewall you're looking for. For a full firewall on either a dedicated system or one of your own, I cannot strongly enough recommend Astaro Linux firewall. Better throughput than a pix, worlds easier to operate and configure, and comparable in price. Very SIP/VoIP friendly. Loads of optional modules (we use its mail filter module to filter spam/viruses for several hundred thousand user mailboxes, for instance) to limit the cost to what you need. Also has a built in SIP Proxy, although I've never used it. Excellent platform. Of course, at home, I just use a little Linksys WRT box. It's hardly a corporate-grade firewall, but it's quite SIP-friendly. N. No votes for Vyatta? I have been seriously checking it out. Thanks, Steve T I played with a demo of Vyatta and it looks pretty good. We've been using mostly Endian (www.endian.com) or M0n0wall. I've had good luck with both of those. Darren Wiebe dar...@aleph-com.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Messaging System
Ricardo Melendez wrote: Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Only for voice messages not SMS. Exists some application based on Asterisk that makes this, or any code to implement in dialplan Thanks in advance. Ricardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We've released an application to do that on www.callblast.org. -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zopier Client
Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg I've been using it on my notebook. I've been happy with it but I'm not a heavy user. The biggest reason I purchased a few copies of it is that I need to have several different sip and iax2 connections for testing purposes. -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inexpensive device for bandwidth management
My thoughts were similar. Availability has not been a problem for us on the WRT54GL boxes. We're pulling them out of our wholesaler all the time without any problems. Darren Wiebe dar...@aleph-com.net Jeff LaCoursiere wrote: And why not DD-WRT, which runs on many more platforms including some more recent platforms still selling on shelves? :) j On Sun, 5 Apr 2009, Mike wrote: I just reread my question and realized I might not have been clear enough. What I meant is that it only seems to works on older Linksys hardware revisions. How do I make sure I can get those? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Sunday, April 05, 2009 15:30 To: oliv...@hh174.be; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Inexpensive device for bandwidth management Actually that was my original thought. BUT?according to what I read on their FAQ, the hardware that can be used is rather limited. How do I secure a reliable supply of those? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hh174 Sent: Sunday, April 05, 2009 14:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive device for bandwidth management Linksys (cisco)WRT54GL and the tomato firmware. 5 minutes setup Olivier Mike a ?crit : Thanksthe thing is I need many device (one for each of my hosted customers) and I'd like this process to be as easy for non-techies as possible, because some of those are technologically-challenged, and need to install the box by themselves or with the help of an IT person that only knows how to install a run of the mill router. So an out-of-the-box thing would be better, but I was recommende the pfsense before and will take a look at it. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Sunday, April 05, 2009 13:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive device for bandwidth management The following two links deal with the same familly of boxes. Generally it's $20 for a case, $20 for a powersupply, but you've probably got an old one that will work. and almost all of their boards are under $200, except for the ones with lots of gigabit interfaces. Many are under $100. http://www.mikrotik.com/ http://routerboard.com/ On Sun, Apr 5, 2009 at 11:07 AM, Mike mailto:l...@virtutel.ca l...@virtutel.ca wrote: Hi, I'm looking for a good network device that does bandwidth management. It can be integrated in a router or stand-alone, but must be SIP-friendly. I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest hardware revisions can't downgrade to the version that worked well) and the DI-724GU (SIP-friendly, but bandwidth management is automated and not configurable enough for my taste), both from D-link. What else is out there and allows me to do upstream QoS on cable/DSL links? Both D-Link routers were under 200$ (99$ and 159$ respectively) and were perfect price-wise for my target customers. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Einhorn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDD FULLL
Just restarting it won't do anything. You could use the following command to find any files over 200mb on the system. Be careful about blindly deleting stuff though *find / -type f -size +200M Darren Wiebe dar...@aleph-com.net * David @ULC wrote: I have 320 GB SATA HDD. When I checked my phpsysinfo, it shows 95% HDD is filled. [r...@vicidialnow mailto:r...@vicidialnow ~]# df Filesystem 1K-blocks Used Available Use% Mounted on /dev/sda2 301924504 285002780 1337472 100% / /dev/sda1 101086 11062 84805 12% /boot tmpfs 1553832 0 1553832 0% /dev/shm [r...@vicidialnow mailto:r...@vicidialnow ~]# du 16896 . You have new mail in /var/spool/mail/root [r...@vicidialnow mailto:r...@vicidialnow ~]# df -i Filesystem Inodes IUsed IFree IUse% Mounted on /dev/sda2 77922304 528483 77393821 1% / /dev/sda1 26104 34 26070 1% /boot tmpfs 219910 1 219909 1% /dev/shm You have new mail in /var/spool/mail/root But my concern is how to solve it I even tried restarting the server , though it will kill unwanted process and will release the space but no ho ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please help test the gender detection module at 575-613-4392
Asterisk Asterisk wrote: You have some good points. Justin Newman isn't exactly someone we don't know. However I only I agree that my name wasn't clear, but I was trying to avoid getting a bunch of spam myself. I'm not sure if I've personally ever spammed the list and I'm pretty supportive of the community. I have been part of these lists for many many years. * The message starts by asking you to call a number. That was the help needed and it worked. There have been more than 500 different callers now and they keep coming in. I'm going to need help with a second round of testing, after I release the updates today and Sunday, but I haven't figured out how to entice people to test again. I thought about doing an outbound call and most people probably wouldn't care, but I'm anti-spam myself and that sounds like spam to me! Any thoughts? -- Snipped -- I'll be happy to try it again to see if I've become a male yet. :) Darren Wiebe dar...@aleph-com.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI script
sorry to but in, but...
1 on first line make sure it is 3!/usr/perl/bin not
#!/user/perl/bin
I'd suggest instead: #!/usr/bin/perl
;)
2009/2/21 Yawar Hadi yawarh...@gmail.com
hi steve,
plz make some cahnges and now i have tested it its working fine to me
1 on first line make sure it is 3!/usr/perl/bin not
#!/user/perl/bin
2 change reading variable from get command to argument as
#$no1=$AGI-get_variable('callerid');
$no1=$ARGV[0];
and where u calling scrip cahnge it like exten
=112,1,AGI(Dial.pl|${EXTEN});
hope it will work for u if any problem then reply me ..
i think we have time difference so thats way i quit when u replied .
On Sat, Feb 21, 2009 at 11:17 AM, Yawar Hadi yawarh...@gmail.com wrote:
hi,
Steve i just loged in and go through all replies.
yes its mistakenly written as user instead of /user on first line let be
go through the problem of not reading the vairable and reply back to you
soon.
wait and dont lose your interest .this is the way to learn some thing new
.wait let me to check it ...
On Sat, Feb 21, 2009 at 5:09 AM, Steve Edwards asterisk@sedwards.com
wrote:
On Sat, 21 Feb 2009, michel freiha wrote:
the script is running smoothly but it seems that it's not reading
variables correctly fro asterisk as you can see below:
Maybe the original author would care to step in? I seem to have lost
interest.
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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--
Best regards
Yawar Hadi Noshahi
QAU Islamabad
(+92-0300-5504798)
--
Best regards
Yawar Hadi Noshahi
QAU Islamabad
(+92-0300-5504798)
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[asterisk-users] Detecting which party initiates a hangup
Hi, I would like to know if it is possible to detect which party initiates a hangup - and if so, how this is done. In my asterisk log, I see something like the following: Feb 18 04:14:13 VERBOSE[17488] logger.c: -- Executing Hangup(IAX2/ToHK1-16, ) in new stack Feb 18 04:14:13 VERBOSE[17488] logger.c: -- Hungup 'IAX2/ToHK1-16' This tells me when the call was terminated, but doesn't tell me which party actually hung up first. Is this possible to detect? thanks, Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please help test the gender detection module at 575-613-4392
Pretty cool. I'm almost offended though as I'm not usually guessed as a female of the species. :) Darren Wiebe dar...@aleph-com.net Asterisk Asterisk wrote: Steve, Tried to test and got call could not be completed as dialed. Were you able to connect? If not, please try again. Call volume has been growing. How about a moving stress variable that could be used as a lie detector of sorts or just to measure how certain parts of a script, or certain questions may This is possible. Do you want to call or e-mail to discuss? I guess to get a baseline, you would have to ask a few inert questions. Yes, I definitely need to do this and will probably add this in for the next release. Justin Newman nt_jnewman at yahoo.com *From:* Steve Totaro stot...@totarotechnologies.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Wednesday, February 18, 2009 10:57:47 AM *Subject:* Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro stot...@totarotechnologies.com mailto:stot...@totarotechnologies.com wrote: On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk nt_aster...@yahoo.com mailto:nt_aster...@yahoo.com wrote: This module detects gender and approximate age range. I'm working on getting it's accuracy to 80%+ on a consistent basis, after implementing filters to remove background noise and other artifacts. It's designed for a number of things. To start, I have several clients (primarily mobile content and servers providers) that want to profile and generate demographics of their users for selling advertising. They also want to understand their user base. Plus, some customers have found that male and female users tend to respond differently to different prompts, flows, etc. This helps in designing a system that meets needs of many different types of users. Of course, there are many other uses and I'm sure people can generate some cool ideas. Let me know how it works when you try the test number at 575-613-4392. Also, let me know if you have any interest in the module. Justin nt_jnewman at yahoo.com http://yahoo.com *From:* Ron Joffe ron.jo...@gmail.com mailto:ron.jo...@gmail.com *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Cc:* Asterisk Asterisk nt_aster...@yahoo.com mailto:nt_aster...@yahoo.com *Sent:* Monday, February 16, 2009 11:05:24 AM *Subject:* Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 That's an interesting module. Care to elaborate on what you designed it for ? Thanks, Ron On Monday 16 February 2009 13:29, Asterisk Asterisk wrote: I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others call too. Also, let me know if you have any need for the module. Justin Newman nt_jnewman at yahoo.com http://yahoo.com Tried to test and got call could not be completed as dialed. This sounds very interesting Justin. -- Thanks, Steve Totaro Justin, how about building some additional functionality. How about a moving stress variable that could be used as a lie detector of sorts or just to measure how certain parts of a script, or certain questions may prove to be more stressful where simply rewording them may have a less stressful response? I guess to get a baseline, you would have to ask a few inert questions. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- Bandwidth and Colocation
Re: [asterisk-users] Looking for SIP loud ringer
We've done this with good results. You can also get one that flashes a bright light for not a lot of money. Darren Wiebe dar...@aleph-com.net Steve Gladden wrote: If you wanna go low tech. down dirty you could also go with a conventional POTS phone line 'loud ringer' device and simply hook it to an ata such as a PAP2, and add the PAP2 to the ring group. Why don't you put a PC in the storeroom with a softphone to be the loud ringer? You could make the ring though the speakers be as loud as the system would support. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, January 28, 2009 9:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Looking for SIP loud ringer Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Psssst - hey buddy, wanna' get a job? (follow-up to asterisk-biz please)
Folks, First of all, this email is sent to -users and -biz, but please follow- up to the -biz list only. I have set the reply-to, but I fear mailman will strip it off ... Please don't flame me for posting to -users, I'm just not sure who lives on -biz as the (signal/noise | net.kook | troll) factor has been pretty bad on that list lately (poor Rehan!!) ;-) Telephony Depot (www.telephonydepot.com) is always looking for talented people who get excited about open telephony platforms such as Asterisk (in its various forms), FreeSwitch, Yate etc, and who just plain find telephony hardware strangely fascinating ;-) Despite the media's recent report that VoIP is Dead* (are you kidding???) we now know from Don Witt** that the economic downturn will actually benefit VoIP (seriously Don, how did you make those magic graphs??). Looking inward at our own sales numbers, it would seem to suggest that there was a startling dip as the financial carnage was first breaking, but that the inimitable can-do entrepreneurial spirit that goes with emerging (occasionally disruptive) technologies is finally breaking through the doom and gloom, and sales are once again back on track. *phew* To help us keep up with demand, we've decided to add more sales, support and programming depth to our bench and have immediate need for a few people in our Philadelphia office. We're looking primarily for: - inside sales rep - entry-level tech support engineer (must grok asterisk company) - web developer / programmer (ideally with solid javascript and/or XML/ SOAP experience) We're a young growing company (Inc. 500 two years running and one of the top 10 private companies in Philadelphia in 2007) and we're definitely open to any other strategic hire that might help take us to the next level of growth. If you think you have something to contribute to online merchandising of hardware around open telephony, send your resume and a brief introduction to w...@our-domain-name (sorry email harvesters!). Telecommuting may be an option for some positions, so if Philadelphia is a non-starter please don't assume this rules you out! Remember, in the unlikely event you want to reply to this email on- list, please follow-up to the -biz list only. Otherwise, please email privately/directly off-list to the email address above (*hint* - this is your first test). Sincerely, -- Darren Nickerson Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) * http://www.fiercevoip.com/story/skype-voip-dead/2008-09-17 * http://voxilla.com/2009/01/19/is-the-2009-voip-surge-theory-correct-1065___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift installation problems
What version of Asterisk and what version of app_swift?
On 29 Oct 2008, at 15:10, [EMAIL PROTECTED] wrote:
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo make install i get:
if ! [ -f /etc/asterisk/swift.conf ]; then \
install -m 644 swift.conf.sample /etc/asterisk/swift.conf ; \
fi
if [ -f app_swift.so ]; then \
install -m 755 app_swift.so /usr/lib/asterisk/modules ; \
fi
and when i do just sudo make, it spits out a ton of junk, this is at
the end:
/usr/lib/gcc/i486-linux-gnu/4.2.4/include/stddef.h:214: error:
declaration for parameter ‘size_t’ but no such parameter
app_swift.c:451: error: expected ‘{’ at end of input
make: *** [app_swift.o] Error 1
Im not sure whats going on here, i have setup asterisk and gotten it
configured with the x-lite soft phone, so i know that is working. I
am ultimately trying to use adhearsion to integrate with my rails
app. I have also installed cepstral voices and these work in the
terminal so i am confident that is also installed correctly.
Thanks.___
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Re: [asterisk-users] Current Open Source Billing Package
Jerry Jones wrote: After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? tia Jerry Have a look at astpp (www.astpp.org) along with OSCommerce. This should do what you're looking for and you do not need to link to Asterisk, etc. Darren Wiebe [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door phone
Not sure if this counts as affordable but: http://www.voipsupply.com/cyberdata-voip-intercom -Darren On Mon, Oct 27, 2008 at 8:46 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Mon, Oct 27, 2008 at 8:36 AM, hbk [EMAIL PROTECTED] wrote: Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Is it a phone or just a box with a button? If it is a phone you could just use an ATA. If it is a box, maybe the usual suspects like VoIP supply have something or Teledynamics is almost guaranteed to have something (but they are a wholesaler). -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and voice recognition
Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and voice recognition
Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php Not sure how the implementation works with Asterisk but I know it's been done (I'd google it). - D On 26 Oct 2008, at 20:55, Christian wrote: Hi, Many thanks for that info. Is there any free solution available as well? Many thanks, Christian On 2008-10-26 at 20:32 Darren Sessions wrote: Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interpreting Asterisk Logs
Hi, Can anybody point me to an online resource that will assist with interpreting Asterisk log files? I note that a similar question was asked in this forum some time ago (http://lists.digium.com/pipermail/asterisk-users/2007-June/189793.html), which doesn't appear to have received any responses. On that occasion, the OP was seeking a log parser - I'm looking for more of a general reference guide. I'm quite new to Asterisk, and VOIP in general, and I'm struggling to understand what many of logged messages mean. The current approach I am taking is to google for specific messages (or parts thereof) - and this has been somewhat fruitful, if not quite tedious. It would be nice to have a reference guide that lists the most common log messages, and what they mean. Does such a guide exist? thanks, Darren -- DOCOMO interTouch provides a full suite of integrated solutions to the hospitality industry. With over 1000 employees operating in 63 countries, DOCOMO interTouch is one of the world's largest hotel technology service providers, backed by mobile communications leader NTT DOCOMO. Email disclaimer - www.docomointertouch.com/Email_Disclaimer.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Well, after very quickly making a test call it's not Vitelity. It could be something with your account? Might want to try opening a support ticket. If you want, create a sub account and e-mail me off list the username and password and I'll test it with my box or vice versa. On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I believe. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Interesting, I've been using them since April and haven't had a problem. I know they changed their server settings a while back but didn't notice anything recently. On Tue, Oct 7, 2008 at 11:47 AM, Roderick A. Anderson [EMAIL PROTECTED]wrote: Darren Severino wrote: Well, after very quickly making a test call it's not Vitelity. It could be something with your account? Might want to try opening a support ticket. If you want, create a sub account and e-mail me off list the username and password and I'll test it with my box or vice versa. You might also want to just check your settings at Vitelity. Over the last six months they have changed the server I'm support to connect to two or three times so my * box was not connecting to them. Therefor no service. I've I'd had it up for more than testing, and been testing, I'd have notices if there was any rime or reason for the changes. No notifications even. Rod -- On Tue, Oct 7, 2008 at 10:38 AM, Stephen Reese [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The voicemail command should be Voicemail([EMAIL PROTECTED]) so in extensions.conf exten = 101,n,Voicemail([EMAIL PROTECTED]) As for the console when you launch it add v's to set the debugging level 'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I believe. Just to make sure, you are doing a 'module reload' each time you make changes to configuration files right? Cool I've got voicemail :-). I am reloading it and have increased the logging level. When dialing out I'm seeing: -- Executing Dial(SIP/101-08183018, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-0818b178 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Oct 7 10:34:34 WARNING[6465]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Think it's a problem with vitelity? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vitelity Asterisk configuration help
Stephen, What exactly are you trying to accomplish? If you want basic call
in/out you're just about there. Changes need to be made in your
extensions.conf. Your phones, by default, are in the [default] context. In
other words when making a call it looks for extensions here. To allow
outbound calling include your outgoing context within the default. To
include it, at the bottom of the default context add include = outgoing
either of these should allow outgoing calling. As for incoming, add a Goto
as follows.
[inbound]
exten = 9045622082,1,Answer
exten = 9045622082,n,Goto(default,101,1)
That equates to goto the default context, extension 101, at the 1st
priority which is your Dial command.
Best Regards,Darren Severino
On Sat, Oct 4, 2008 at 1:30 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the full configurations
that are available are a bit complex. Any tips or recommendations to
get up and running would be great.
sip.conf
Code:
[general]
register = rsreese:[EMAIL PROTECTED]:5060
context=default ; Default context for incoming calls
realm=ns1.neocipher.net ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
domain=neocipher.net; Set default domain for this host
[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=test81
mailbox=101
callerid=Stephen 101
host=dynamic
dtmfmode=rfc2833
canreinvite=no
reinvite=no
disallow=all
allow=gsm
[102]
type=friend ; allows incoming and outgoing calls
username=102
secret=test81
mailbox=102
callerid=(Bob 101)
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
allowguest=yes
insecure=very
promiscredir=yes
musicclass=default ; Sets the default music on hold class
for all SIP calls
[authentication]
[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=pass
allow=all
insecure=very
canreinvite=no
[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=pass
allow=all
canreinvite=no
extensions.conf
Code:
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail(102)
exten = 102,1,Dial(SIP/102,20)
exten = 102,2,Voicemail(102)
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) ;This
automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Answer
voicemail.conf
Code:
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes ; Context to Send voicemail from [option 5
from the advanced menu]
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
[default]
101 = 123,Stephen Rese,[EMAIL PROTECTED]
102 = 123,Bob Dole,[EMAIL PROTECTED]
1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk Load Balancing
One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balancing
I know. :) I've already mentioned some of the OpenSIPS options to him on the OpenSIPS users list (LCR module specifically). Just brain dumping everything that came to mind. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 10:31 PM, Alex Balashov wrote: OpenSIPS/Kamailio have modules designed specifically for that kind of functionality now without a need for an outside monitoring process or SRV reliance. Darren Sessions wrote: One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Failover System
I agree that an OpenSER solution on top of Asterisk for a 120 users is massive overkill to say the least. High calls-per-second? Multiple Asterisk servers? Multiple vendors? Advanced LCR? or plans for any of that in the near future? Then I think it makes sense to look at fronting Asterisk with OpenSER for such a small amount of users. Asterisk can do everything you'll need it to do otherwise. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 1, 2008, at 7:44 PM, Alex Balashov wrote: Jai Rangi wrote: Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) All depends on how important those 120 users are. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco + Asterisk
Any particular reason you're using H323 instead of SIP ? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 16, 2008, at 12:04 PM, Guilherme Loch Waltrick Góes wrote: I have a Cisco 3845 with a ISDN PRI port connected to my legacy PBX, this router is running IOS 12.4(5) T5. I'm trying to integrate Asterisk with this router through H.323, I tried ooh323 (comes with asterisk-addons) and it works partially, I can make calls from Cisco to Asterisk, but the other way around dosn't work. Does anybody have any hints of what could be wrong ? -- Guilherme Loch Góes Notícias e Fórum sobre VoIP com software livre: http://www.voipexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX?
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for this particular application. An OpenSIPS solution will take care of your traveler's NAT issues (and could handle the registrations) while you used Asterisk for voicemail and whatever else. I've personally used this type of general setup in the past with a great deal of success for remote offices and soft-phones on laptops. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 9, 2008, at 1:19 PM, Mattias Andersson wrote: Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy for their own phones (Running Widows XP). The reason is that the are using laptops and travel, some are already using softphons and IAX bout some don't like softphons for some reason. If it is not any proxy out their, the will I write o of my own. (Of cause giving it out for free), I think Asterisk for Windows would be overkill. Sorry for my poor English. Regards Mattias Andersson Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk phone conferencing performance
You shouldn't have any delays at all. Are you using ztdummy for timing? and what kind of load does the box have on it? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 9, 2008, at 4:23 PM, George Williams wrote: Hi, I just set up my first Asterisk with MeetMe conference support on my local LAN. It works great, but I think it may need a little tuning - I am getting audio delays of up to 1 second. Should I expect better performance in this area, or is this to be expected? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf programming?
A cheaper alternative would be the voip wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Sep 4, 2008, at 12:13 PM, Mark Michelson wrote: Ken D'Ambrosio wrote: Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm afraid I've forgotten a fair bit. One big thing that I've forgotten is the syntax, etc., for extensions.conf. Where do I find that? I'm looking for stuff about commands, syntax for commands, variables, etc. Is there a handy-dandy manpage, webpage, or what-have-you that I'm missing? Thanks! -Ken Your best bet is to read chapters 5 and 6 of Asterisk: The Future of Telephony. Here's a link for the book itself: http://www.oreilly.com/catalog/9780596510480/ Here's a link for the downloadable pdf: http://downloads.oreilly.com/books/9780596510480.pdf Here's a link for the book in html format http://tfot.leifmadsen.com Best of luck to you! Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines inAGI
Impressive work Bradley! I tested it and it worked great, even with my
mandatory 'use strict'.
Thanks,
- Darren
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 29, 2008, at 5:47 AM, Watkins, Bradley wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Sessions
Sent: Thursday, August 28, 2008 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Tips and Tricks: Dynamic
Subroutines inAGI
...
The hurdle in doing something like this was how to
dynamically execute
a subroutine from the results of the database query which
were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds /
knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution
(without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number,
1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
If you don't mind disabling strict refs (no strict 'refs';), you
could easily do this.
This would allow you to use something like: $sub($argument1,
$argument2);
The only other way I can think of (though I have not tried it) would
be to populate a hash with subroutine refs and use the string as the
index into it.
Something like this:
#!/usr/bin/perl
use strict;
use warnings;
sub print_ref { print @_; };
my %sub_hash = (print_ref, \print_ref);
sub print_stuff {
my $sub = shift;
my $string = shift;
$sub($string);
}
print_stuff($sub_hash{print_ref}, This is printed.\n);
The first idea uses the symbol table directly, and the second one
essentially is building your own symbol table.
Hope that helps,
- Brad
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Re: [asterisk-users] Reliable wireless SIP phones
On Aug 28, 2008, at 9:06 AM, Jaap Winius wrote: Hi list, Are there any reliable wireless SIP phones available on the market yet? We typically prefer DECT in which case the SNOM M3 is a strong contender, but recently our customers have told us good things about Polycom's new wifi handset: http://www.telephonydepot.com/product_p/105-058-8002dual.htm One limitation is that there's no minibrowser, so you won't be able to navigate the http proxy signup/authentication page at your local coffee shop. Works great in the typical office setting though! Sincerely, -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines in AGI
When I set out to develop a basic service provider Perl AGI framework
for Asterisk three or four years ago, I wanted to design something
that would make developing additional Perl AGI apps under this
framework scalable and easy to do. One of the features I wanted to
have in this framework was the ability to do a database dip on a
particular incoming called number to see which service I needed to
execute and then to dynamically execute that subroutine from the
database servers results. I could switch services or point the number
to a canceled operator message by simply doing an update to that
telephone number’s record in the database - instantly re-provisioning
the telephone number.
The hurdle in doing something like this was how to dynamically execute
a subroutine from the results of the database query which were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds / knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution (without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number, 1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
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Re: [asterisk-users] Pri to sip interfaces
Are you using an Asterisk PBX? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:06 PM, Tom Moore wrote: Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pri to sip interfaces
You can use an extremely simple Asterisk config to do the SIP-PRI call conversion that'd be very solid. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:37 PM, Tom Moore wrote: No, these are mainly Samsung pbx systems. I know I can use Asterisk to do this but what be a solid platform to go with that can go in the phone closet? tom From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Darren Sessions Sent: Wednesday, August 27, 2008 9:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pri to sip interfaces Are you using an Asterisk PBX? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:06 PM, Tom Moore wrote: Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] implementing an intercom with asterisk
For simple paging the bogen tamb works very well. Just hook it up to an fxs port and you're good to go. Darren Wiebe [EMAIL PROTECTED] Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Thanks, and if this isn't appropriate for this list, if anyone has a better destination for the question, Id be quite appreciative. -j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail has issues with DTMF
If the Linksys unit is forced to a single specific DTMF type, and Asterisk is set specifically to something other than the Linksys, then when the Asterisk server answers the line your DTMF will not be recognized. If your outbound termination vendor supports the Linksys' DTMF settings, then that would also explain why outbound PSTN DTMF is functional. Hope this helps. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 23, 2008, at 12:39 AM, Max Alex wrote: Hi everybody, I have linksys phone at my location, i am using asterisk version 1.4.19, I have a issue regarding dtmf mode, i have set the Asterisk DTMF mode to Auto in order to eliminate Asterisk effect on the DTMF transmission. Both Inband and AVT from Linksys worked with PSTN IVR. But, We have the issue why Asterisk Voicemail doesn't work with Linksys set to Inband and Asterisk set to Auto. And what is the reply of asterisk while the dtmf configuration like this? Anyone please help me for this issue, i have searched many pages but i haven't found the exact solution or reason for this? -- Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT Satellite?
I've used C-Band, Ku-Band, and DVB satellite internationally with VoIP for years at a previous employer and rarely had any problems was the sat link was up and running. If you do plan on having 'remote offices', you'll want to make sure they all come back to a central earth station (hub and spoke topology) or you'll have virtually insurmountable latency issues (as Femi mentioned). Whatever you do though, don't stick the remote offices with their own internet bandwidth using VPN to connect to the home office for voice, data services as VPNs are extremely problematic over satellite. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 23, 2008, at 4:45 PM, Femi wrote: I’ve used VOIP over satellite for years and while it’s not perfect it is sometimes actually better than cellular voice Unless you have a double hop scenario where the traffic makes two satellite hops from one remote to a central hub and then to another satellite remote the latency is actually not noticeable Satellite usually has a latency of 250 – 300 ms and in most cases this does not have a noticeable effect on the conversation Femi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Tom Moore Sent: 23 August 2008 15:50 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Semi-OT Satellite? Hi, using Asterisk over satellite can be done. Not all satellite providers are created equal and some are better than others. If you are going to do communications between offices that are connected over satellite office to office you may have a problem. My personal choice for satellite connections is the Idirect platform. Tom From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Ken Williams Sent: Saturday, August 23, 2008 9:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Semi-OT Satellite? We're entertaining moving our intranet to Hughes satelite for our remote locations. I'm curious if anyone with Asterisk servers has used satellite, and if so, is the latency an issue. My understanding is that you immediately introduce 250ms latency for travel time up and back down, however it is a much more direct connection then offered by traditional land lines. Perhaps someone has some other suggestions? We've started looking into Global Crossing as an alternative to have more control and reliability between all of our remote facilities, maybe this is a better alternative. Our biggest problem is most of our sites are in smaller cities where your bigger connections are more limited. Looking for any suggestions. Thanks, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set callerid with plus sign
Just change your dial command and add the plus sign there. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 1:28 AM, ronald wrote: Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the same thing. is this possible? thank you Regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
It's tough to say why a voice would start sounding like a robot. There are so many variables that could effect your Asterisk server. I always go for process of elimination when I have a problem similar to this with call quality. What I would do is install an end point on the same local network / subnet as your asterisk server (either a hard phone or a soft phone like X-Lite by Counterpath). Register the phone locally with your Asterisk server and make some calls or put an echo tester up. If things sound good, you know your Asterisk server is working just fine, and the problems lies somewhere on your network between the Asterisk server and whatever gateway / device. If it sounds awful, and the codecs match, then it's time to start troubleshooting the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 3:14 AM, bilal ghayyad wrote: Dear Darren; You might be right because one day it happened with me and the situation was same like this as following: The status that the ping result is very good for all partied (Asterisk machine, IP Phones on the Internet), and no problem in the processor utilization or RAM or hard disk space. Previously, we changed the DSL router and it worked fine !! But what can I do on the Asterisk level to overcome the problem? I already enabled the jitter on the IAX and SIP, but did not resolved. And I am using the G729 codec and sometimes I use GSM. Any advise for the robot voice with weak battery :) ?! Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED] Date: Thursday, August 21, 2008, 9:47 PM I doubt recompiling is going to help you unless you've got a very unstable system (hard drive going out or something), and then you've got bigger things to worry about then anyways. You should install (if you haven't already) the 'top' program. Top gives you a nice set of system statistics and a list of processes. If you're only having issues on the IP origination side of things, I would start checking your bandwidth and latency on your network. Is the originating end point on the Internet? or local? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:55 PM, bilal ghayyad wrote: Dear Darren; I discovered that calling from the Asterisk to the IP Phone Extension (like calling from mobile to digium and then enter the IP Phone extension, or calling from fxs to the IP Phone extension), it goes very good without any problem. But calling from the same IP Phone to another IP Phone or to any mobile (via fxo port) or to the fxs, it cause the problem (voice become very very bad, like robot with weak battery or sick man). Another way for the problem, if I called from another Asterisk PBX to our Asterisk PBX (that has the problem) and the call was via IAX, and I was need to reach to the IP Phone, then I hear the voice like robot with weak battery. So, the problem appear if the call originator was IP and not TDM. What could be the reason for the problem? No one did any change, I am sure, it suddenly become like this. Any help? Regards Bilal --- On Thu, 8/21/08, Darren Sessions [EMAIL PROTECTED] wrote: From: Darren Sessions [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non- Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, August 21, 2008, 6:13 PM I'd run top on the server to see if the CPU utilization is going through the roof. If you use AGI, make sure there aren't any orphaned processes consuming resources. If all else fails on the software side of things, I'd restart the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote: Hi All; My asterisk version is 1.4.19.2 and it contains one digium card of 2 fxs and 2 fxo ports, it was working great for more than one month without any problem. Suddenly, any call will be done, then voice becoming like robot (or sick man), it slow and cutting. I restarted the machine, but it is the same !!! I checked the RAM which is 1 GB and I found a lot of space. Any advise what could be the problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Problem with modem data calls and xorcom astribanks
Not sure what you've heard before, but I have successfully used a modem at 9600 baud (forced via AT commands) through a zaptel card on several occasions. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 8:14 AM, Greg Woods wrote: I have been told before on this list that a modem through a zaptel card will not work. And mine doesn't, at least not for data calls (it works fine for fax). Apparently the modem requires the full bandwidth of the POTS line, which you do not get through the zaptel card. You might at least check to make sure that echo cancellation is turned off. That can interfere with a data call. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] After Dial execution, using DIALEDTIME, ANSWEREDTIME
We recently discussed DeadAGI on the list - I'd check the archives
first.
I just finished doing a write up on DeadAGI and Perl on my website if
you're interested.
DeadAGI *can* be very reliable if done properly.
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 21, 2008, at 9:35 AM, selmak se wrote:
Hi,
I noticed that when dial terminates it does not return to the
dialplan, and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME,
ANSWEREDTIME if I can not execute anything after Dial()?
I made a workaround with DeadAGI (below) but it is unreliable: if 2
calls end nearly at the same time I do not know to whom belongs the
ANSWEREDTIME value :
exten = h,1,DeadAGI(myagi.agi,0,${DIALEDTIME},${ANSWEREDTIME},00)
Any comments?
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Re: [asterisk-users] Suddenly the voice become like robot (cutting), like sick man
I'd run top on the server to see if the CPU utilization is going through the roof. If you use AGI, make sure there aren't any orphaned processes consuming resources. If all else fails on the software side of things, I'd restart the server. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 21, 2008, at 4:03 PM, bilal ghayyad wrote: Hi All; My asterisk version is 1.4.19.2 and it contains one digium card of 2 fxs and 2 fxo ports, it was working great for more than one month without any problem. Suddenly, any call will be done, then voice becoming like robot (or sick man), it slow and cutting. I restarted the machine, but it is the same !!! I checked the RAM which is 1 GB and I found a lot of space. Any advise what could be the problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Perl AGI defunct process
Ruddy,
I've used deadagi for years with perfect success.
If it's a perl agi module, you need to make absolutely sure that
you're using 'use strict' and 'use warnings' in the main agi file -as
well- as any includes. You'll need to test your agi while in console
mode, so any of the perl warning messages that get sent to the console
are visible. You'll want to get rid of any errors and warnings.
In addition, I've setup my agi scripts to execute cleanup functions
when they detect any kind of sig message just for good measure.
$SIG{INT} = 'cleanup';
$SIG{TERM} = 'cleanup';
$SIG{QUIT} = 'cleanup';
$SIG{HUP} = IGNORE;
With this approach, as I said before, I've ran perl agi apps in very
high call volumes at various companies for years without any issues.
Hope this helps.
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 19, 2008, at 10:20 PM, Igor A. Goncharovsky wrote:
Hi!
Ruddy Gbaguidi wrote:
I'm using DeadAgi and has set AGISIGHUP to no because I don't want my
script to stop if the user hangs up.
But when it reach the end of the script, the child process should
die.
And I don't see why I only have this trouble with perl agis.
Can you check if your script realy don't get SIGHUP?
Some time ago I have problem with that setting AGISIGHUP to 'no'
have no
effect.
--
Best regards,
Igor A. Goncharovsky
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Re: [asterisk-users] US-based echo test servers?
Another thing you may want to do is try a simple ping test to the far end host. While this may not always be a reliable way to test lag given that the far end maybe just a proxy and your RTP may be terminating to another device, it still should give you a good idea what your lag times are at least on the signaling end of things. You could also do a traceroute to see how many hops your having to jump through as well. You could use a tool like ngrep to actually see the sip signaling and copy out the media gateway from the SDP if you really wanted to, and do a ping on that. I've done extensive work with international voip origination and termination, and typically I haven't had any problems unless it's going over satellite (lag) or there is a problem at the far end (usually pdd or quality issues). If things keep up, I'd even consider running top during a call to see what kind of utilization your local server is at just to make sure something isn't wrong there either. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 18, 2008, at 10:41 AM, Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open door automatically...
Set it so when they dial the number, it calls an AGI script that instantly answers and generates a call file and hangs up. That way, you could dial and then hangup, and the system generates a call file that calls the door phone and does whatever it needs to do separate of the initial call. I just posted a Perl based call file generator to the list not to long ago that would easily work for this application. Hope that helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 13, 2008, at 4:20 PM, Carlos Chavez wrote: I have a new setup that uses a 2N Entrycom door phone that has a switch to open an electric lock. The way this works is that when you are speaking with someone at the door you dial a code and it releases the lock on the door. This part works great. My customer wants to be able to dial a certain number and have the door open automatically without having to wait on the phone. I can simulate this option by using the D option of the Dial command to send DTMF to the door phone once it answers. The only problem is that they do not want to wait until the door phone answers. They just want to dial a number and hangup immediately. How can I do this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dialer proof of concept
Here is a simple Perl implementation to generate call files . .
You'll still need something for it to execute after the call files are
generated; either a simple AGI app that streams a file, a Macro, or a
nice dialplan layout.
In any case, you could call something like this very rapidly with
whatever parameters to create as many call files as you felt like, and
Asterisk would start acting on them immediately (if the call files
were generated without wait time).
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_
use strict;
use warnings;
sub call_file_name_generator {
my ($len, $str, @chars);
$len = shift;
@chars = ('a'..'z','A'..'Z','0'..'9','_');
foreach (1..$len) {
$str.= $chars[rand @chars];
}
return($str);
}
sub call_file_generator {
use Asterisk::AGI;
my ($channel, $retries, $retry_interval, $wait_time, $application,
$data, $ob_clid) = @_;
if (!$channel || !$retries || !$retry_interval || !$wait_time || !
$application || !$data || !$ob_clid)
$AGI-verbose(Missing data to create call file!!, 1);
return(1);
}
my $ob_file = /var/spool/
asterisk/.call_file_name_generator()..call;
unless(open(CFILE, . $ob_file)) {
$AGI-verbose(Can't open call file for writing!!, 1);
return(1);
}
$file = \#\nChannel: .$channel.\n\nMaxRetries: .$retries.\n;
$file.= RetryTime: .$retry_interval.\nWaitTime: .$wait_time.\n
\n;
$file.= Application: .$application.\nData: .$data.\nCallerid:
.$ob_clid.\n;
printf CFILE $file;
close(CFILE);
system(mv $file /var/spool/asterisk/outgoing);
return(0);
}
On Aug 8, 2008, at 1:48 PM, Bradley Sumrall wrote:
I am a returning Asterisk user.
It has been a few years since I played with it and trying to get a
server up for proof of concept
What is the easiest method of having asterisk dial 5 numbers
simultainiously and deliver a pre recorded message?
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Re: [asterisk-users] Transcoding
I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub- versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I am a **BIG, BIG** fan of OpenSUSE. :) Use yast under 'Software Management' and do a search for 'gsm'. Make sure gsmlib and gsmlib-devel are *both* installed. Then scroll down and make sure that libgsm and libgsm-devel are *both* installed. After that, you'll have to recompile Asterisk. See if that does anything for you. - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 8:48 AM, Guilherme Loch Waltrick Góes wrote: I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is pretty bad. The softphone is in the same LAN as the Asterisk server, so I don't think it's a bandwidth issue. Best Regards, On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions [EMAIL PROTECTED] wrote: I would make absolutely sure you've got your linux distro's version of libgsm installed. I can't really speak to the difference between those two versions of Asterisk without looking at a change-log, but I highly doubt a serious modification to the gsm code took place between sub-versions. Hope this helps, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding
I have used virtually all versions of Asterisk 1.0+ (literally, either in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel and haven't had any issues with gcc optimizations with regards to audio sounding choppy. This scenario for me has always been the gsm libs. _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 6, 2008, at 9:16 AM, Mark Michelson wrote: Guilherme Loch Waltrick Góes wrote: I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some prompts recorded in GSM format. I have these same prompts in another server with Asterisk 1.4.18, on this server the prompts sound pretty nice, but on the first one they sound pretty choppy. Was there any changes on the transcoding code between this 2 versions ? Any hints ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br One important difference between the servers may be the compiler used. We have heard reports that using GCC 4.2 or later with optimizations on causes choppy audio when using GSM. Solutions to this include either downgrading your compiler to GCC 4.1 or earlier, or selecting DONT_OPTIMIZE in menuselect under compiler options and then recompiling Asterisk. I also believe that you can set the optimization level for compilation to -O2 in Makefile.rules and have no choppy audio, but I cannot confirm this. Of course, if this server isn't running GCC 4.2, then you can ignore everything I've said so far :) Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Least Cost Routing
ASTPP (www.astpp.org) will do calling cards / prepaids as well as lcr. Darren Wiebe [EMAIL PROTECTED] emist wrote: Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
I can speak first hand to this having gone through it just a few months ago . . After being spoiled with all the features and standard compliance in Postgres, I was put in a position with a new project to setup a redundant (Master-Slave) database cluster. I immediately jumped to Postgres to do the job (using 8.3). My biggest gripe at the time was that there was really nothing built IN postgres to do the replication as I soon found out. Everything was third party and there were several replication modules suggested to me that seemed stagnant or un-maintained or required an older version of Postgres (bypassing the massive performance increase of the 8.3 release). Of those that I did try that were opensource, all of them seemed fairly complex to get up and running - to say the least. Also having used MySQL extensively, I decided to give it a test run on a separate set of boxes. I'm not exaggerating when I say the replication was up and running in about 10 minutes. While I do appreciate (a lot) how standards compliant Postgres is, MySQL was an absolute clear winner in my book with regards to the replication. Just my two cents . . - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 3, 2008, at 12:26 PM, Tzafrir Cohen wrote: On Sun, Aug 03, 2008 at 08:13:30AM +0100, Grey Man wrote: We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution Hmmm... is that really the case? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_flite 0.6 released
I've updated the app_flite module to work with the Asterisk 1.6.x code- base in addition to it already working with the 1.4.x, and 1.2.x. (1.0.x support is untested and unsupported). It can be downloaded on my website at: http://www.darrensessions.com/downloads/app_flite-0.6.tar.gz Additional details are in the ChangeLog and README files in the tar ball. As always, if there are any questions or comments, please forward them to me at [EMAIL PROTECTED] Thanks, - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
If you had a dax in front of all your circuits, you could move them from one server to another without physically touching anything. I've done about 300 calls on a dual processor box doing just SIP with an entirely AGI based setup and it held up just fine, but doing TDM, I'd worry about your PCI bus at those call levels. - D _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 1, 2008, at 1:07 PM, Al Baker wrote: You mean running , 400 Calls on 1 BOX ? Even if you COULD do it, the gods of TELCO would have you burn in hell for stacking that much critical traffic on ONE Intel, non - high availability box Jerry Geis wrote: Assuming you have a Quad core machine, at least 4 GIG ram, will a machine like this handle 4 Quad T1 cards? is that advisable? What about running AGI's on such a machine. Will the machine handle starting/stopping all those AGI's? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginner Issues
I had issues like this on one installation that cleared up when I turned ACPI and APIC?? off in bios. Darren Wiebe [EMAIL PROTECTED] Gerard A. Matthew wrote: Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from my T-Mobile BlackBerry Handheld ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ** app_swift v1.6.2 released for Asterisk 1.6.x code-base **
2008-07-08 - app_swift v1.6.2 released for Asterisk 1.6.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args are specified Can now wait for DTMF after text-to-speech processing is done if the timeout and max digits args are specified Entire DTMF input placed into channel variable Can be downloaded from http://www.darrensessions.com Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ** app_swift v1.2.2 released for Asterisk 1.2.x code-base **
2008-07-09 - app_swift v1.2.2 released for Asterisk 1.2.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args are specified Can now wait for DTMF after text-to-speech processing is done if the timeout and max digits args are specified Entire DTMF input placed into channel variable Can be downloaded from http://www.darrensessions.com I promise, this is the last release notice. :) Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ** app_swift v1.4.2 released for Asterisk 1.4.x code-base **
2008-07-08 - app_swift v1.4.2 released for Asterisk 1.4.x code-base --- Added support for handling multiple dtmf input Added support for input timeout and max input digits (similar to AGI's get_data) Ignores DTMF if no timeout and max digits args are specified Can now wait for DTMF after text-to-speech processing is done if the timeout and max digits args are specified Entire DTMF input placed into channel variable Internal cleanup Can be downloaded from http://www.darrensessions.com In addition, an Asterisk 1.6.x code-base version is almost complete. Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adit 600 password reset
Are you trying ethernet or serial? Have you tried the other? -Darren From: [EMAIL PROTECTED] on behalf of C F Sent: Thu 5/22/2008 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Adit 600 password reset In the manual it's mentioned that if the DIP switch marked RST is set then it will reset CLI password. I have not been successful in doing that. Has anyone tried it? I bought one off eBay and can't get in because of username password that I don't know. I am assuming local is set to off. Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime status feature - user feedback needed
Just FYI, I wrote an application that tracks the status of SIP or IAX2 extensions by listening to the AMI. It was for use by callshops but would probably require minimal change to work for you. It's currently part of the ASTPP source code. Darren Wiebe [EMAIL PROTECTED] Atis Lezdins wrote: On Thu, May 8, 2008 at 3:49 AM, Ex Vito [EMAIL PROTECTED] wrote: On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. True, that was one of initial options, however I prefer to NOT have yet another layer. I will consider this as an option where appropriate. However this looks quite awkward to me, somehow it reminds me tailing queue_log or CDR and putting result into MySQL database.. just one level more that way. For now, I see only one point against this - having status cleared upon module load/unload makes it easier to follow restarts/module loads. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! Ok, so we're exactly at the point. Yes, I agree that it would act nearly the same way as AMI actions, however there's one great advantage - It would be really easy to set this up for user. AMI proxy would take more effort, need configuration, etc. Then there should be much more development support for proxy than for code within asterisk (if you have noticed, there's no new code, just reusing existing functionality) I think that there should be several ways how to do something, not just one. Having realtime status won't mean that much changes, for now I can see only 4 families for this - queue_members (already existing), queue_callers, channels and meetme. Really nothing more to give full overview of Asterisk Status. Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan, Extensions, etc. Worksheet
If you're willing to cc me a copy I'll be in your debt. Thanks, Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Has anyone created a worksheet they can share for designing a dialplan, extensions, voicemail, etc. I'm making my way through the O'Reilly Book (dead tree version) and finding it enlightening. I have hacked at dialplans created by others but never actually came up with a design for my own system. It's sort of a work in progress made of bits and pieces from all over. Having a real plan would probably make things easier. Rod -- Rod, You will be glad that you are taking the learning curve plunge down the road. No pain, no gain. I can certainly say that I am glad I got into Asterisk way before there was any real documentation or GUIs for that matter. It forced me to learn the real deal Asterisk through trial and error which is invaluable if you plan on really getting into it. Then again, if you want easy, use a GUI. Easy isn't what I'm after. I was hoping for planning worksheets. Something to go over with a customer (I know I said this was for my personal system but that is the first step). How many extensions/ phones/ softphones, and what their /numeric/ extension will be. An IVR plan and the text that goes with it, voice-mail handling and mailboxes, etc. This type of stuff. So from the minimal number of responses -- yours :-) -- I'm going to guesstimate no one has anything like this at all or that they can or are able/willing to share. Out comes the notepad and the thinking cap. /-| Cheers, Rod -- Thanks, Steve Totaro Hey Rod, I think I may be able to help with worksheets from 3com, NEC, and other system vendor's sales channel. It obviously will not match exactly to Asterisk but will give you a great foundation for the functions and features that you need to question. I have my own but I prefer not to put it in the public domain. It is adapted from a conglomeration of many different proprietary systems that I have dealt with. I think many others have the same and consider it proprietary internal information for their business. Let me see what I can dig up from my archives. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] prepaid on the trunks
Am I correct in thinking that one application of this would be monitoring what you have left for funds with a prepaid vendor? Darren Wiebe [EMAIL PROTECTED] Brian J. Murrell wrote: On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Wouldn't you just equate a Calling Card (that's the unit that has an account balance and charges against it) with a trunk instead of a user or extension? You can call the astcc agi script with any value you want for a Calling Card identifier. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] prepaid on the trunks
Ok, I'm not aware of this feature in astcc and I can't speak for astbill or a2billing. I do know that I coded it into astpp and it's called vendor rating in there. It works but it's not used a lot at present. Darren Wiebe [EMAIL PROTECTED] Nhadie Ramos wrote: hi sir, yes that would be it, but instead of having a prepaid provider, i will setup my own as5300 and asterisk will talk to that. is that possible in astcc, astbill or a2billing? regards, nhadie Am I correct in thinking that one application of this would be monitoring what you have left for funds with a prepaid vendor? Darren Wiebe [EMAIL PROTECTED] Brian J. Murrell wrote: On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Wouldn't you just equate a Calling Card (that's the unit that has an account balance and charges against it) with a trunk instead of a user or extension? You can call the astcc agi script with any value you want for a Calling Card identifier. b. */Nhadie Ramos [EMAIL PROTECTED]/* wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Regards, Nhadie On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote: i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this? TBH, I don't really understand your description, but I will say that I implemented astcc a week or two ago and it works for what I need. Cheers, b. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift v1.6.1 released for Asterisk 1.6
Thought I'd let everyone know I've released app_swift v1.6.1 which is entirely based off of Will Orton's work he's placed in the public domain. Works great with Asterisk v1.6.0-beta7.1. In any case, can be downloaded from my site at: http://www.darrensessions.com Go easy on me, this is my first release of anything. Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
I've used Adit600's almost exclusively for my installs. All have worked great for me. -D From: [EMAIL PROTECTED] on behalf of Steve Totaro Sent: Thu 4/3/2008 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks. Just Google Quintum Tenor AX. Well worth the money. Thanks, Steve Totaro On Mon, Mar 31, 2008 at 10:03 PM, Al lists [EMAIL PROTECTED] wrote: Im guessing T1cas not PRI,just because its giving 24 fxs per T1. Steve, what are my options for SIP to fxs? thank you! On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentPBX mirror?
CentPBX has bit the dust I believe. -D From: [EMAIL PROTECTED] on behalf of Chris Bagnall Sent: Wed 4/2/2008 7:12 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CentPBX mirror? Greetings list, Not exclusively asterisk-related, but I've noticed the CentPBX site has been offline the last few days. Anyone know the reasoning behind that, and more importantly, is anyone mirroring it? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it http://www.minotaur.it/ This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
Yup.Trixbox. -D From: [EMAIL PROTECTED] on behalf of Al Baker Sent: Sat 3/29/2008 2:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question How did you chose Centos, versus Red Hat, Suse, Debian, ? Was there some key feature it offered that the others didn't ? Cost ? Darren Wright wrote: Notifications can be done either thru SNMP traps or SMTP. Insight Manager is free from HP, but any SNMP trapper can work with alerts. The recovery CD is just a build that reloads the majority of the system with a static ip. We backup off site to one of our servers via FTP. ILO access is an integrated IP KVM. So you can see the machine boot, get virtual media access, etc. O/S is CentOS. For smaller systems, RAID 1, and for larger DL380 based systems 0+1 -D -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Al Baker Sent: Thursday, March 27, 2008 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! Darren Wright wrote: One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non- techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
Notifications can be done either thru SNMP traps or SMTP. Insight Manager is free from HP, but any SNMP trapper can work with alerts. The recovery CD is just a build that reloads the majority of the system with a static ip. We backup off site to one of our servers via FTP. ILO access is an integrated IP KVM. So you can see the machine boot, get virtual media access, etc. O/S is CentOS. For smaller systems, RAID 1, and for larger DL380 based systems 0+1 -D -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Al Baker Sent: Thursday, March 27, 2008 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! Darren Wright wrote: One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non- techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non-techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID T1 PRI
That's not going to tell you anything about the digits in transit. That's just telling you that your PRI is up. you are going to need exten = 4DIGITS From: [EMAIL PROTECTED] on behalf of broadband Voice Sent: Sat 3/15/2008 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID T1 PRI Additional output [EMAIL PROTECTED] ~]# /sbin/ztcfg -vv Zaptel Version: 1.4.9 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: Clear channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels to configure. On 3/15/08, broadband Voice [EMAIL PROTECTED] wrote: Can you share with me sample extensions.conf? This is what I have exten = 215xxx,1,Dial(Zap/1) in zapata.conf [channels] context=external switchtype=ni1 resetinterval=3600 overlapdial=no priindication=outofband facilityenable=yes signalling=pri_cpe usecallerid=yes cidsignalling=bell hidecallerid=no restrictcid=no usecallingpres=yes echocancel=yes callerid=asreceived faxdetect=incoming nsf=sdn group=1 channel=1-23 zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 On 3/15/08, Darren Wright [EMAIL PROTECTED] wrote: Feel free to ping me off list. I've setup quite a few Cavtel PRI's with *.the paperwork they asked you to setup? Typically, they only send 4 digits. Do you have the questionnare they asked you to fill out? dwright at d2 - tech dot com. From: [EMAIL PROTECTED] on behalf of broadband Voice Sent: Fri 3/14/2008 9:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID T1 PRI Thanks. I am in Philly. I may have to configure the extensions.conf well to pass the incoming channels. On 3/14/08, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice [EMAIL PROTECTED] wrote: I had Cavalier turn up a T1 PRI. How can I put in the DIDs to direct to Asterisk. Here is a log Zaptel Tool (C)2002 Linux Support Services, Inc. ⤠T2XXP (PCI) Card 0 Span 1 ââ[3;10Hâterfaces â â[3;37Hâ â â â â â âCurrent Alarms: No alarms. â rd 0 Span 1 â â âSync Source:T2XXP (PCI) Card 0 Span 1 â rd 0 Span 2 â(R) â âIRQ Misses: 0 â â â âBipolar Viol: 0 â â â
Re: [asterisk-users] DID T1 PRI
Feel free to ping me off list. I've setup quite a few Cavtel PRI's with *.the paperwork they asked you to setup? Typically, they only send 4 digits. Do you have the questionnare they asked you to fill out? dwright at d2 - tech dot com. From: [EMAIL PROTECTED] on behalf of broadband Voice Sent: Fri 3/14/2008 9:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID T1 PRI Thanks. I am in Philly. I may have to configure the extensions.conf well to pass the incoming channels. On 3/14/08, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Mar 14, 2008 at 8:13 PM, broadband Voice [EMAIL PROTECTED] wrote: I had Cavalier turn up a T1 PRI. How can I put in the DIDs to direct to Asterisk. Here is a log Zaptel Tool (C)2002 Linux Support Services, Inc. ⤠T2XXP (PCI) Card 0 Span 1 ââ[3;10Hâterfaces â â[3;37Hâ â â â â â âCurrent Alarms: No alarms. â rd 0 Span 1 â â âSync Source:T2XXP (PCI) Card 0 Span 1 â rd 0 Span 2 â(R) â âIRQ Misses: 0 â â â âBipolar Viol: 0 â â â âTx/Rx Levels: 0/ 0 â â(R) â âTotal/Conf/Act: 24/ 24/ 0 â â â â 112 â â â â123456789012345678901234â Back â â â â âTxA â â â âTxB â â â âTxC â â âTxD â â â â14Câ âRxA â Loop â â â Quit â â âRxB â â âRxC â â âRxD â ââ â â ââ T2XXP (PCI) Card 0 Span 1 F10=Back I need to add 215-xxx- etc to come in to the Asterisk box. Do you have DIDs already? When you call a DID and watch the Asterisk console with a little verbose, you should see the call come and how many digits the telco is sending. Then you need to make matching entries for those DIDs either in the form of exact matches or pattern matches to do pretty much whatever you can imagine. Are you in Philly? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experiences with grandstream GXW 4024 FXS?
Faraz Khan [EMAIL PROTECTED] wrote: Dear all, Just wanted to know if any one had deployed the Grandstream GXW 4024 yet. Wanted to hear any feedback and/or problems with this unit that you may have experienced. Faraz, I'd be surprised if you get many responses to this the 4024 hasn't been released to distributors yet. I think we're still a couple of weeks out on that product. Sincerely, -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra phones and park/pickup feature
You'll want to use the XML park and pickup with the aastras. Feel free to ping me off list if you need help. -Darren Dwright at d2-tech dot com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Monday, March 03, 2008 2:45 PM To: 'Asterisk Users List' Subject: [asterisk-users] Aastra phones and park/pickup feature We are installing Aastra phones (480's and 57i's) into a fairly simple asterisk setup. Although call park pickup work fine using xfer to 700 (to park), dial 701 (to pickup), we are unable to make the park/pickup softkey feature work on the aastra's. Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is setup like this: softkey4 type: park softkey4 label: Park softkey4 value: asterisk;70 softkey4 line: 1 softkey4 states: connected softkey4 type: pickup softkey4 label: Pickup softkey4 value: asterisk;70 softkey4 value: 1 softkey4 states: idle, outgoing (we also tried asterisk;700 with the same result). Has anyone got the softkey park/pickup working on aastra? Thanks Michelle This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
I've used lots of Digium T1 cards on DL380 / DL320's without a hiccup. -Darren From: [EMAIL PROTECTED] on behalf of Joshua Kinard Sent: Tue 2/26/2008 5:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Norman Franke Sent: Tuesday, February 26, 2008 5:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Had it with Dell Garbage On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote: On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? HP DL380 is my baby. Thanks, Steve Totaro Ditto. We've been using HPs for a while without problem. I'm currently using a DL380 (a recent quad processor one) and it screams. -Norman This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID providerforCanada, USA and Europe
Yup, SIP is working ok as well, except for the cross-country 100ms round trip. Their answer was to upgrade to 1.4 Not an option for me. Please ping me off list so we can further discuss. dwright at d2 - tech dot com -Darren From: [EMAIL PROTECTED] on behalf of John Faubion Sent: Sun 2/24/2008 1:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Suggestions for reliable DID providerforCanada,USA and Europe I've had some serious issues with Teliax as of late with their new Denver server. DTMF issues, IAX2 connection issues, and major latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility issues. I have had zero problems with their old servers. Interesting... I've got several lines on Teliax that have been in place for several months and the service has been very good. Recently we connected a new system to Teliax and I've been fighting the same issues you mention. I've been told the problem is with my software since SIP seems to work fairly well but not IAX. I also found out that my system is one of the first 20 systems to connect to their new Denver server. Now I'm curious about how many others are having the same problem. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID provider forCanada, USA and Europe
I've had some serious issues with Teliax as of late with their new Denver server. DTMF issues, IAX2 connection issues, and major latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility issues. I have had zero problems with their old servers. Voicepulse has been WAY better, but no flat charges, no 729. Frankly, even my broadvoice (yikes!) connection has been significantly better, no 729. For a full Virutal PRI, I'd look at a provider that can give you the port and SIP connections, like XO. I've had good success with XO's product. -Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Saturday, February 23, 2008 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Suggestions for reliable DID provider forCanada, USA and Europe I used TelIAX for a while and was happy with the service. I used it for testing before we connected to our PRI... http://www.teliax.com On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote: I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which echo-can for Digium B410P ?
The HWEC, not software. -Darren From: [EMAIL PROTECTED] on behalf of Olivier Sent: Thu 2/21/2008 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Which echo-can for Digium B410P ? Hi, Which echo-canceler shall I pick for Digium B410P ? Is HPEC relevant ? Reading from its datasheet, it seems related to analog cards. Regards This message was sent from D2 Technology, INC. winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog DID
An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price.Darren Wiebe[EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DIDOn Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks?What is an analog DID trunk?You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500This is for providing plenty of analog extensions (phones). Is that whatyou're after?-- Tzafrir Cohenicq#16849755 jabber:[EMAIL PROTECTED]+972-50-7952406 mailto:[EMAIL PROTECTED]http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir___-- Bandwidth and Colocation Provided by http://www.api-digital.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog DID
Hey, that's cool! I wish I'd known that 6 months ago.Darren Wiebe[EMAIL PROTECTED]Wed Feb 13 2008 10:33:31 AM MST from James Finstrom to Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Analog DID-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Rhino's Analog cards support analog DID. no need for all the extrastuff You will want to get an R8FXX with fxs modules that will giveyou channels in sets of 2.ADID has not really taken off in the OS telephony market I think dueto a lack of understanding people stay with the proprietary phonesystems that pimp this feature. Okay so I will take the lead and pimpit for asterisk. With Rhino Analog cards you CAN do ADID with no extraequipment. However if you want to spend the money we can go the otherroute :)darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. Darren Wiebe [EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DID On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b327c1163231152562594! -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b327c1163231152562594!- --James FinstromRhino Equipment Corp.Tel: 1-800-785-7073 ext. 6344FAX: +1 (480) 961-1826IP: asterisk.rhinoequipment.com ext 6344FWD: 633686 ext 6344THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARYMATERIAL and is thus for use only by the intended recipient. If youreceivedthis in error, please contact the sender and delete the email and itsattachments from all computers.-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.6 (GNU/Linux)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKBGxd6H7YOdzXfygVuBygzAw===51QY-END PGP SIGNATURE-___-- Bandwidth and Colocation Provided by http://www.api-digital.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
